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Fix 1v1 screenshare audio stopping on mic mute #31263

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Fix 1v1 screenshare audio stopping on mic mute #31263

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6ea99f6
Fix 1v1 screenshare audio stopping on mic mute
paterkleomenis Sep 7, 2026
61eebfe
Update lib_webrtc for 1v1 screenshare mic-mute fix
paterkleomenis Sep 7, 2026
d40e71a
Update lib_webrtc to combined mixing-audio PR
paterkleomenis Sep 7, 2026
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32 changes: 29 additions & 3 deletions Telegram/SourceFiles/calls/calls_call.cpp
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Original file line number Diff line number Diff line change
Expand Up @@ -520,7 +520,12 @@ rpl::producer<Webrtc::DeviceResolvedId> Call::captureMuteDeviceId() {

void Call::setMuted(bool mute) {
_muted = mute;
if (_instance) {
if (_screenWithAudio && _screenAudioControl) {
_screenAudioControl->setMicrophoneMuted(mute);
if (_instance) {
_instance->setMuteMicrophone(false);
}
} else if (_instance) {
_instance->setMuteMicrophone(mute);
}
}
Expand Down Expand Up @@ -1187,9 +1192,13 @@ void Call::createAndStartController(const MTPDphoneCall &call) {
sendSignalingData(bytes);
});
},
.createAudioDeviceModule = Webrtc::AudioDeviceModuleCreator(
saveSetDeviceIdCallback),
};

_screenAudioControl = std::make_shared<Webrtc::MixingAudioControl>();
descriptor.createAudioDeviceModule
= Webrtc::MixingAudioDeviceModuleCreator(
Webrtc::AudioDeviceModuleCreator(saveSetDeviceIdCallback),
_screenAudioControl);
if (Logs::DebugEnabled()) {
const auto callLogFolder = cWorkingDir() + u"DebugLogs"_q;
const auto callLogPath = callLogFolder + u"/last_call_log.txt"_q;
Expand Down Expand Up @@ -1493,6 +1502,13 @@ void Call::toggleScreenSharing(
}
_videoCaptureDeviceId = QString();
_videoCaptureIsScreencast = false;
if (_screenWithAudio && _screenAudioControl) {
_screenAudioControl->setMicrophoneMuted(false);
_screenAudioControl->setLoopbackEnabled(false);
if (_muted.current() && _instance) {
_instance->setMuteMicrophone(true);
}
}
_screenWithAudio = false;
if (_systemAudioCapture) {
_systemAudioCapture->stop();
Expand All @@ -1507,6 +1523,15 @@ void Call::toggleScreenSharing(
_videoCaptureIsScreencast = true;
_videoCaptureDeviceId = *uniqueId;
_screenWithAudio = withAudio;
if (_screenAudioControl) {
_screenAudioControl->setLoopbackEnabled(withAudio);
if (withAudio && _muted.current()) {
_screenAudioControl->setMicrophoneMuted(true);
if (_instance) {
_instance->setMuteMicrophone(false);
}
}
}
if (_videoCapture) {
_videoCapture->switchToDevice(uniqueId->toStdString(), true);
if (_instance) {
Expand Down Expand Up @@ -1695,6 +1720,7 @@ void Call::handleControllerError(const QString &error) {
}

void Call::destroyController() {
_screenAudioControl = nullptr;
_instanceLifetime.destroy();
Core::App().mediaDevices().setCaptureMuteTracker(this, false);
if (_systemAudioCapture) {
Expand Down
2 changes: 2 additions & 0 deletions Telegram/SourceFiles/calls/calls_call.h
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Original file line number Diff line number Diff line change
Expand Up @@ -37,6 +37,7 @@ namespace Webrtc {
enum class VideoState;
class VideoTrack;
struct DeviceResolvedId;
class MixingAudioControl;
} // namespace Webrtc

namespace Calls {
Expand Down Expand Up @@ -387,6 +388,7 @@ class Call final
std::vector<not_null<PeerData*>> _conferenceParticipants;

std::unique_ptr<tgcalls::Instance> _instance;
std::shared_ptr<Webrtc::MixingAudioControl> _screenAudioControl;
std::shared_ptr<tgcalls::VideoCaptureInterface> _videoCapture;
QString _videoCaptureDeviceId;
bool _videoCaptureIsScreencast = false;
Expand Down
2 changes: 1 addition & 1 deletion Telegram/lib_webrtc
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Submodule lib_webrtc updated 2 files
+972 −0 webrtc/webrtc_create_adm.cpp
+36 −0 webrtc/webrtc_create_adm.h
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